1. Field of Invention
The invention relates generally to an arrangement and a method for measuring and assessing properties of a system which transfers an electrical, mechanical or acoustical signal, or converts such a signal into another signal. The system is characterized to have at least one signal input and at least one signal output. Examples of such systems include an electro-acoustical transducer (loudspeaker, actuator, head phones), a converter between the analog and digital domain, storage media for audio data (CD, mini-disc), and wired and wireless communication systems (fiber optics, high frequency transmission).
2. Description of Related Art
A signal which is converted, transferred or stored in a system can be subject to distortion caused by properties of the system (e.g. inherent nonlinearities) and their interaction with the transferred signal. Additional stochastic disturbances may be caused by noise, ambient sound, or loose connections, which are not directly related to the transferred signal. Traditional techniques developed for assessing the signal distortion typically require providing a special excitation signal (single tone, multi-tone complex), measuring the output signal of the system, and transforming the time signal into the frequency domain to search for additional components in the output spectrum which are not part of the excitation signal. This technique makes it possible to identify harmonic and intermodulation components at multiples of the excitation frequencies, and at any combinations of the difference and sum frequencies. Distortion measures standardized by national and international organizations assess the amplitude of the distortion components, whereas the phase of the distortion component is neglected. The second-order and third-order distortion, total harmonic distortion and other simple measures are sufficient in most cases. For example, these measures are commonly used to assess the effects of regular loudspeaker nonlinearities (motor and suspension nonlinearity, nonlinear radiation) which are directly related to their principle of operation.
A conventional signal distortion measurement system of this type is shown schematically in FIG. 1, which provides a traditional measurement of signal distortion generated by a system under test 1 by using a spectral analysis (FFT). This technique may be applied if the stimulus contains a limited number of tones and each distortion component may be separated from the fundamental tones and identified as a harmonic or intermodulation component. Typically, this method uses a signal generator 2 which generates a single tone, a sensor or measurement input 4, an analog/digital converter (ADC) 6, a FFT analyzer 8 and a block 10 for calculating the relative distortion dt in percent.
However, there other types of signal loudspeaker distortion which are quite audible, but which can not be reliably detected by traditional measures. These kinds of distortion are mainly caused by anomalies and defects caused by problems in design or manufacturing. For example, loudspeakers may have defects such as a loose glue joint producing a buzzing sound, a voice coil rubbing on the pole tips, or any obstacle hitting the moving assembly and generating a small click. This class of signal distortion is called “triggered distortion” because it is deterministic; i.e., it depends on the input signal and is initiated under special conditions of the state variables (e.g. voice coil displacement). This triggered distortion may produce significant peak values for a short time in rare instances. However, the power in the mean is much smaller than that found with regular distortion caused by motor and suspension which has a steady-state characteristic. Performing a spectral analysis (FFT transform) is not a reliable way to detect triggered distortion because the energy of the triggered distortion is distributed over a large number of higher-order harmonics (>40), and the signal to noise ratio of each component is very low.
U.S. Pat. No. 5,884,260 discloses an invention which addresses this problem by measuring the envelope of the time signal using a filter bank; this approach is illustrated in FIG. 2. A signal generator 12 generates the stimulus for the system under test 14. A sensor or measurement input 16 provides its output to a filter bank 18, which contains multiple branches connected in parallel. Each branch comprises a band-pass filter 20, a rectifier 22 and a low-pass filter 24 connected in series. The pass bands of the band-pass filters and the time constants of the low-pass filters correspond with properties of the human auditory system. The band-pass filters have sufficient damping outside their pass-bands to separate the fundamental components from the harmonics. The amplitude and phase response of band-pass filters 20 and the time constants of the low-pass filters 24 changes the waveform of the analyzed signal and limits so as to detect signal distortion which is short in duration but high in amplitude. This method provides a pattern of the distortion which is relevant to human hearing, but is not comparable with other measurements and is hardly interpretable from an objective point of view.
The techniques known in the prior art fail if the triggered distortion or the symptoms of a malfunction or defect have less power than the measurement noise or the regular distortion caused by normal nonlinearities inherent in the system without any defects.